Sampled amplitude read channels are commonly employed in recording devices, such as optical and magnetic storage systems, for detecting and decoding binary data stored on a disc medium. A transducer (read head), positioned in close proximity to the disc, senses alterations on the disc's surface such as magnetic transitions or optical "pits", which represent the recorded binary data. The surface alterations induce a corresponding change, or pulse, in the analog read signal emanating from the read head. Thus, the read channel must detect and translate these pulses into an estimated binary sequence which, in the absence of errors due to system dynamics, will be the originally recorded binary sequence.
In sampled amplitude read channels, the read signal is equalized into a predetermined partial response (e.g., PR4 or EPR4) meaning that the response of the channel to an isolated surface alteration (i.e., an isolated pulse) will take on a particular shape. The output of the channel can then be approximated as a linear combination of time delayed pulses modulated by the binary input sequence; for example, a binary "1" translates to a positive or negative pulse (alternating) and a binary "0" translates to no pulse.
An important advantage of partial response system is the ability to control and compensate for intersymbol interference (ISI) from neighboring pulses, thereby allowing an increase in recording density without compromising the bit error rate (BER). This intersymbol interference is taken into account when the pulses are detected using a maximum likelihood sequence detector, such as a Viterbi detector, comprised of a state machine, or trellis, "matched" to the particular partial response target. The sequence detection process entails sampling the analog read signal and evaluating the sample values in context. That is, a predetermined number of consecutive input samples are compared to a number of valid binary output sequences, taking into account ISI, and the most likely binary sequence in euclidean space is selected as the correct output sequence. Errors in equalizing the read signal to the desired partial response, however, can cause errors in the detection algorithm (i.e., it can cause the wrong output sequence to be selected) because the sequence detector is no longer "matched" to the channel's response.
Thus, to optimize operation of a sampled amplitude read channel, the filter(s) responsible for equalizing the read signal must be adjusted to an optimal operating setting. This is normally accomplished through some type of "off-line" filter calibration procedure, or through a "real-time" adaptive equalization algorithm, or a combination of both. For example, the filters may first be optimized off line to determine initial optimum settings, and then the filters may be fine tuned in real time during normal operation using, for example, a least mean square (LMS) adaptation algorithm.
The equalization function in a sampled amplitude read channel is typically carried out by a front-end analog filter for equalizing the analog read signal before sampling, and a downstream discrete filter for equalizing the sample values before sequence detection. The analog filter also performs an anti-aliasing function by attenuating noise caused by under sampling the analog read signal.
A prior art method for calibrating the equalizer filters "off line" is to read a test pattern recorded on the disc and compute a mean of squared sample errors (MSE). The test pattern is read several times using a number of different filter settings, and the setting that minimizes MSE is selected as the initial operating point. Thereafter, a real time adaptation algorithm may fine tune the filter settings during normal operation.
There are problems associated with calibrating the equalizer filters not addressed by the prior art. Namely, when a recording system is first powered on after manufacturing, the dynamics of the system are unknown. Consequently, the equalizer filters must be set to "nominal" settings which, hopefully, will allow the read channel to read a test pattern recorded on the disc and execute the filter calibration routines. This "nominal" setting, however, may not be sufficient to enable the read channel to successfully read any data from the disc--thereby disabling the calibration process and the read channel altogether. This undesirable result is more likely to occur due to process variations in the analog components, such as the analog equalizer. That is, the nominal setting for one analog equalizer may be vastly different from that of another. When the process variations are significant, it becomes impracticable to search for an optimal setting by retrying the calibration routine with different filter settings until an operable setting is found.
Furthermore, calibrating the analog equalizer in real time (i.e., with an adaptive algorithm) is typically too complex and expensive to implement. This relegates the analog equalizer to operating settings determined by the off line calibration procedure--where the off line calibration is preformed intermittently (e.g., only at power on) to avoid degrading the storage system's overall performance. Thus, if the system dynamics change during normal operation due, for example, to temperature drift, such changes are not normally accounted for in the analog equalizer.
What is needed, then, is a way to calibrate the analog equalizer in a sampled amplitude read channel without having to rely on "nominal" settings to enable an off line or real time calibration procedure. That is, there is a need to calibrate the analog equalizer without having to read any data from the disc. Further, there is a need for a sampled amplitude read channel capable of calibrating the analog equalizer during normal operation in order to compensate for changes in the system that can occur over time. Still further, there is a need to calibrate the analog equalizer quickly during normal operation to avoid significantly degrading the storage system's overall performance.